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B&O Tech: 100 Years of Danish Loudspeakers

This book was released some months ago as part of the 100 year anniversary of loudspeaker development in Denmark. There are a number of good articles in there, including a historical article on the importance of loudspeaker directivity at Bang & Olufsen.

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27 Responses on “B&O Tech: 100 Years of Danish Loudspeakers”

Bartsays:

What a beautiful book on Danish audio, and what better way to celebrate Peter L. Jensen’s invention by putting the current best loudspeaker in the world on the cover.
I personally think that when Peter L. Jensen is looking down from heaven and sees the Beolab 90 he’s smiling.

Bart have you heard a Beolab 90 yet? Where? I don’t believe it can be the best loudspeaker in the world because it does not yet have a correct radiation pattern. Most of its sound is cast in the forward direction, which would be the opposite of the live model. With live sound there is a small amount of direct sound, followed by a wide early reflected field and finally the full reverberant field. In the reproduction, we need to cast more sound in the reflecting direction and position the speakers to take advantage of the huge, spacious reflected field that is possible. This makes the reproduction sound fields resemble the live sound more closely and makes the speakers disappear as sources of sound themselves.

These principles are really quite simple, but surprising enough that they may not stumble upon them for another 100 years because audio engineers would not normally think of comparing the image model of reproduced sound to that of live sound, or using more reflected than direct sound in the radiation pattern.

Firstly, the Narrow Beam Width setting of the Beolab 90 is only one of its 7 possible directivity settings. If you prefer to have more reflected energy, then this is easily possible by using the remote control to switch the loudspeakers to “Wide” or “Omni” Beam Width. So, it seems to me that dismissing the entire loudspeaker due to one possible setting of its many parameters is to make a judgement without nuance.

Secondly, I don’t understand your insistence that the purpose of audio reproduction is to create a “live sound”. The recording engineer and the mastering engineer (or, in the case of a film, the re-recording engineer) make decisions regarding aspects of the recording (including spatial attributes) experienced at the listening position using a pair of loudspeakers in a given acoustic environment. The purpose of reproduction for a listener is to reproduce the experience that those persons intended to convey when they made the recording.

Of course, if you want a different sound field than the one on the recording, then one way to achieve this might be to use the acoustical behaviour of the listening room to add this effect on the recording (another way would be to play your recordings through a reverb unit, as other people propose). However, this will, as one example, result in the frequency-dependent Inter-aural Cross Correlation being either partially or primarily defined by the listening room rather than by the Interchannel Cross Correlation of the recording. In other words, you are applying a spatial filter to the recording within the playback system. As a rather silly, extreme example, of you play Suzanne Vega’s recording of “Tom’s Diner” on a pair of loudspeakers placed in the Cathedral of St. John the Divine in New York, you will hear lots of spaciousness and reverb – but I would argue that if that spaciousness and reverb was desired by the people who made the recording, they would have added it themselves.

I don’t suggest that you should not prefer a wide-dispersion loudspeaker and sound systems that deliver lots of reflected energy. I just argue that this is your preference (and is certainly also the preference of some other people with whom I have had similar conversations) and not a Universal Truth. Since this is a potential preference, the Beolab 90 includes the option of a wider directivity – while still maintaining as constant (or “frequency-independent”) a directivity as possible / feasible.

I am glad that they have included an omni mode – if true – I don’t know if there are any drivers on the back of the unit – but omni is still too hot on the direct sound. This can work OK in a very large room, but not in most domestic sized rooms where you are so close to them. Remember, if the speakers themselves are the strongest output in the room then they will be obvious sources themselves and detract from the spaciousness that is possible.

There is no sound field until the recording is played back, and there is no IACC contained in the recording. Both ears are free to hear both (or all) speakers, and crosstalk is a part of the system. It is the speakers and room that create the sound fields in the reproduction, not the recording.

In a nutshell, most speaker manufacturers think that stereo is two speakers beaming their sound into your two ears, where your brain then is “fooled” into making stereo. The room treaters and other experts will even try to damp out all reflections so that you might have the privilege of hearing only that precious direct sound from the speakers. This is a great tragedy and has been going on now for most of the stereo era, to the detriment of all of us.

But NO – stereo is a field-type system in which the function of the loudspeakers is not to be direct radiators but rather to be image model projectors, creating sound fields within the listening room that more closely resemble live sound – a reconstruction of the sound fields contained in the recording rather than some mistaken binaural confusion. What you should hear is an auditory scene spread across the whole front of your room, the speakers disappearing as obvious sources, rather than a spread of sounds between the speakers with little depth or spaciousness. I eagerly await an audition of the 90s, but I have never found speakers that are too hot on the direct sound to be able to do this. I have in my home some speakers that were built in accordance with image model theory principles that I could demonstrate if you are ever in Florida. If the engineers are interested I can describe how to build them and use them.

I can see that you and I have a fundamental disagreement about the purpose of an audio reproduction system, and what is and is not contained in a recording. We also have fundamentally different views of what loudspeaker manufacturers (including Bang & Olufsen) assume. Personally, I think that it’s wiser to not bother debating this, since I don’t see either of us convincing the other of our opinions or preferences.

I can also glean from your statements that there are some features of the Beolab 90 that may come as a surprise to you (such as the distribution of loudspeaker drivers and their control in the DSP). Some of these features that I think may interest you (in particular, Beam Direction Control) have not yet been released in the software. Consequently, I would suggest that if you are planning to audition a pair of Beolab 90’s, then you should wait for a while. I can let you know when the Beam Direction Control has been released if you would like. In the meantime, I invite you to read the Technical Sound Guide for the Beolab 90 (available as a PDF, linked from this page), since some of your comments indicate that there are many features of the loudspeaker of which appear to be unaware – and this is colouring your assumptions and opinions about its capabilities.

Please let me know if you’d like a heads-up on the feature releases as they come out. I’ll probably be posting the “big ones” here anyway.

Cheers
-geoff

Bart De Biesays:

Hi Garry,
You claim to have speakers with (according to you) so called live sound?
Because of that (if I may say so) pretentious claim i am curious to hear what speaker brand you are talking about?
What’s the panoramic power response of those speakers?
How closely is the output of a pair of those stereo speakers matched?
What is the dynamic capability/headroom of those speakers?
Are those speakers able to reproduce the 10 octaves of human hearing (20 Hz to 20kHz)?
Did you get a chance to look at some of the links I posted?

The fine tuning for the three front Beam Widths (Narrow, Wide, and Omni) is finished – so these will not change. So, you can hear the finished product now. Note, however, that the Active Room Compensation filters are currently customised for only one of these Beam Widths – whichever one the dealer was using when she/he made the measurements.

Cheers
– geoff

Gary Eickmeiersays:

I didn’t take anything you said as offensive. We audio guys are all a little opinionated and take it into account. My speakers are based on a new theory for stereophonic sound called image modeling in which we try to make the image model of the reproduction sound field as much like that of a typical live sound field as possible. It deploys a matrix of direct and reflected sound into the room the same way as live, except that there are precise positioning requirements to get the whole array to “fuse” into a highly focused soundstage. Most of the principles of the model are contrary to what most audio engineers would think of, so I go around trying to tell about it to jog some more imaginations and directions for research. The speakers were made by a friend in Indiana to my design ideas and meet all of the requirements of loudness and freq response that you ask about. The room is 20 x 30 feet, and the speakers are positioned 5 ft out from the front wall and an equal amount out from the side walls so that the entire matrix of direct and reflected speakers are equidistant from each other. Thus we see that the reflected sound is equal in importance to the direct sound – maybe even more so. It is a sat/sub system with a Velodyne F1800.

Bartsays:

Hi Gary,
Just like you said, we all have our own opinions and of course that’s fine.
I personally find the comments you make about b&o speakers strange since the acoustic lens technology they employ is designed to spread sound horizontally in an equal manner from the lowest to the highest frequencies.
Take the beolab 5 for example: the subwoofer is facing down because sub bass is omnidirectional, the woofer is facing forward because mid bass is quasi omnidirectional, the midrange and tweeter are even less omnidirectional so they are sending their sound into a lens that reflects the sound 180 degrees horizontal.
This in turn creates a speaker that spreads sound as evenly as possible 180 degrees in front of it while the lenses on the midrange and tweeter also minimize ceiling, floor and front wall reflections.
I don’t see the link since this is a very different approach than your direct/reflecting – image model theory speaker technology.
Anyway, each to his own, succes with your speakers and enjoy listening to music on them.

Most of what I said didn’t register with you. I’m not talking about 180 degrees radiation in front of the speaker, I am talking about more sound in the reflecting direction – a negative directivity index -6 dB more output toward the front wall of the room. I recall some of Moulton’s writings saying he didn’t want much sound going toward the front wall, just the side walls, which would explain the differences with his Beolab 5 design. But that is not a correct design. Obviously with live sound the centerfill is just as loud and just as important as the sidewall reflections and the general pattern has to be even all across the soundstage. Speaker positioning is also a major part of the theory and I don’t know what Dave has said on that.

When I said that he has supported me I simply meant that he came up to me in 1989 before I presented my paper and gave me some suggestions and some reading, which I had already done and indicated in my references at the end of the paper. Since then I have written to him but he has not been real responsive, not sure why.

Gary Eickmeier

Bartsays:

Hi Gary,
The beolab 5 was build for a reason.
The lenses on the beolab 5 where designed to transmit treble and mid-range at a horizontal angle of 180 degrees providing the listener with a consistent sonic experience regardless of his/her position in the room. At the same time these lenses minimize floor, ceiling and front wall reflections, why?
The b&o archimedes project provided an important answer to how people perceive sound in a room.
One of the results of the archimedes project proved that sound reflections from floors and ceilings interfere with the human perception of sound, while reflections from walls are seen as a natural part of listening within the confines of a home or studio environment.
The reason we need to minimize front wall reflections is to shorten the sound decay to a point below 50 milliseconds where the reverberance of the playback room never gets a chance to build up.
Reverberance is the part of the playback sound event that carries perceptually audible information about the playback room.
It commences at 50 milliseconds and can run on for many seconds in a large reverberant room.
In small rooms, it usually doesn’t go on for more than a second.
Interestingly, the reverberance that exists between 50 and 150 milliseconds especially tends to interfere with clarity and intelligibility of sound.
Therefore we need to make the end of the room behind the loudspeakers highly absorbent at all frequencies, and the side and back walls highly reflective.
All the energy from the speakers propagates along the length of the room and back and is then absorbed. All early reflections are broadband, and all reverberance is absorbed.
The side walls need to be reflective because laterally reflected energy increases the sense of envelopment from music playback – live or recorded since we need this reflected energy to help localize the sources of sound.
We also want the side walls to have “lateral” symmetry, which is to say that both sides of the room have to be identical in shape and materials, that there is a “median plane” down the middle of the room, and that the two sides of the room constitute “mirror images” of each other. Needless to say, the loudspeakers need to also use this median plane.

Kind regards
Bart

Gary Eickmeiersays:

Thanks very much for that explanation! I tried to talk to Soren Bech during a meeting at the AES about that project, hoping I could have some input, but never connected up with each other. That was a fantastic opportunity to test all kinds of spatial theories about stereo.

What you have just told me has some bad data leading to wrong conclusions. The reflections I am talking about in Image Model Theory are delayed 10 milliseconds from the front wall and slightly more from the side walls after the corner secondary. After that very important second “hit” of the sound (the first reflections, basically) it does not return in any strength to even be able to build up beyond 50 ms.

Also – and very importantly – those side wall reflections should NOT come directly from the speakers themselves as with a 180° dispersion but as secondaries that were launched from the rear radiation of the speakers. The main reason is to be able to control the Direct to Reflected (D/R) ratio and delay. In fact, having too much direct sidewall reflections might cause the undesirable artifact of the whole soundstage collapsing to one side as you move closer to one speaker. I know this from experimentation. I had my prototype speakers with more output toward the side walls in the front radiation, and that is what happened. Then, when I swapped sides with the speakers and they then had more frontal sound cast toward the other speaker, I had a stable center image as I moved about. See Mark Davis’s work on the Soundfield One speaker at DBX. So the lateral reflections must come from the rear radiation and the front wall of the room must be specularly reflective for this to work. You must position the speakers as I have said and you must have more radiation toward the rear by about 6 dB. The speakers are oriented diamondwise into the room, launching the sound from four driver faces, each one facing 45° from the centerline. The rear faces have full gain, the inner front face has -3 dB, the outer front face is -9 dB compared to the rear faces. Final soundstage will be formed from the real and virtual images of all 8 sources equally, the rear ones being “lit up” by the stronger radiation to the rear.

All of these factors are very important to the final result, none of which would normally occur to the speaker designer or experimenter, which is why I am beating the bushes telling my story. I stumbled upon most of it accidentally while experimenting with my 901 speakers some 35 years ago. My new design has normal drivers and a radiation pattern a little different from those, and conforms exactly to my theories about spatial audio and mimicking a live sound field. Please believe me that not having a reflective front wall is a basic error and to try it my way. You have the absorptive end near the rear so that the sound can’t come back to the front after your 50 ms. But the reflected sound from the front wall gives the depth of image and brings the first reflections around to the sides with a proper D/R ratio and delay.

Gary Eickmeier

Bartsays:

Hi Gary,
Stereo was originally developed to be an auditory illusion creating the feeling of being at a live concert.
Humans have 2 ears and thus hear in stereo.
When you are at a live concert listening to music your right ear will receive a sound and so as your left.
Then the brain takes these 2 sounds and processes them so that we appear to be hearing one big soundstage.
The most basic and simple explanation for creating stereophonic sound goes like this:
the right speaker produces the sound our right ear would hear when being at a live concert
the left speaker produces the sound our left ear would hear when being at a live concert
These two independent audio channels, and the signals that are reproduced have a specific level and phase relationship to each other so that when played back through a suitable reproduction system, there will be an apparent image of the original sound source replicating the aural perspective and localization of instruments on a stage or platform experienced during a live concert.
In order for this auditory illusion to properly work we need to make sure that we’re only hearing the sound from the speakers and not the room.
At first glance this seems difficult because we also definitely don’t want to make the room anechoic since humans use laterally reflected energy to help localize the sources of sound.
So how do we do it?
We make the front wall non reflective and the rest of the room reflective!
Because the front wall is non reflective the only front sounds we hear are from the speakers, not the room.
The rest of the walls need to be reflective so that we’re not only perceiving the direct sound energy from the source but as well a volley of early reflections also from the sound source.
We identify this group of sound artifacts as a “single sound” by their unique phase-locked relationship.
These phase-locked volleys of early reflections up to approximately 50 ms. after the direct sound are “fused” together into a single perceptual construct.
From a psychological point of view, then, a “sound” consists of a bunch of phase-locked early reflections that are integrated by the auditory system, rather than consisting of the single direct emission from the loudspeaker.
It is wrong to assume that the front wall of the room should be specularly reflective because if the reflected energy comes from behind or the same direction as the direct sound, it will not support the localization process as well as laterally reflected energy does.
Without sufficient laterally reflected energy, the sound will generally lack a sense of envelopment and musicality since we use these accumulations of sound artifacts to develop an extremely rich perception of the timbre of the sound, its position in the room, our position in the room, and the physical nature of the room (including its dimensions, furnishings and surface materials).

It would not be possible to relate all that is wrong with that explanation in a few paragraphs here. Let me just point out that the right ear does not hear just the right speaker etc, it hears both speakers, and the same with the left ear. This is not an error with the stereophonic system, it is part of it. It is called crosstalk and it marks the dividing line between stereophonic and binaural systems.

In stereo – a field type system – the object is not to record signals for the ears, but rather record and reproduce sound fields in rooms, just like the live sound fields. There can be any number of microphones, channels, and speakers. Binaural, on the other hand, is a system of recording and reproducing signals for the ears on two and only two channels. It is recorded on a dummy head with ears just like ours and is reproduced so that the channels are introduced to each ear separately with no crosstalk and no room sound, either by means of headphones or crosstalk cancelling speakers in a dead room.

The main clue to how to do stereo is to look at the image model of live sound fields. An image model is a plan view of the sources and all of their first and second reflected sound images as additional sources on the other side of the reflecting walls. The live sound model has all of the instruments all across the soundstage emitting direct and reflected sounds from front and side walls. In order to sound the same, the reproduction image model must do the same. It’s as simple as that in the broad stroke.

Gary Eickmeier

Bartsays:

Hi Gary,
It’s true that my explanation of stereo is maybe a bit simplistic and incomplete but the truth remains that it was originally developed to create an auditory illusion where you have two speakers beaming their sound into your two ears, where your brain then is “fooled” into making it stereo.
Ofcourse in a stereo loudspeaker system, sound from the left loudspeaker travels to the left ear, and the right ear – where it arrives slightly later, changed in timbre due to the extra distance, and filtering/equalization effects of wrapping around the face and head.
These delay and eq effects are known as HRTFs, or head related transfer functions.
Similarly, sound from the right loudspeaker travels to the right ear, and also the left ear – again, slightly delayed and changed in timbre.
The sound transfer to each “opposite” ear is known as inter-aural crosstalk, and is an essential and desirable component of a stereo playback or monitoring system.
The reason a sound coming from just the left loudspeaker sounds like it’s coming from just the left loudspeaker (even though the sound travels to both of our ears) is because our brain uses the HRTF sub-millisecond-level time-delay and the eq differences of the wavefront’s arrival at each ear to determine directionality.
We perceive the sound as coming from the direction of the ear at which the wavefront first arrived.
This is known as the law of first wavefront, or Haas effect, or precedence effect.
If both loudspeakers reproduce an identical sound wave simultaneously, both ears receive the same wavefront at the same time, followed by identical slightly delayed inter-aural cross-talk.
The listener perceives this sound as coming from a central location, directly between the loudspeakers – where there is no physical loudspeaker.
The illusion of a sound located where there is no loudspeaker is called a phantom image.
The stereo format is a two-dimensional format when creative recording and mixing techniques are used.
There is clearly a sense of left/right directionality, because it is possible to position sounds anywhere between the loudspeakers, and to create the illusion that sound is coming from just beyond the physical loudspeakers.
It is also possible to create the illusion of front/back depth and to a lesser degree, it is possible to create the limited illusion of a third dimension – height.
I think it’s wiser that we stop debating, since I don’t see either of us convincing the other of our opinions or preferences.
We all have our own opinions and of course that’s fine.
Anyway, each to his own, succes with your speakers and enjoy listening to music on them.

Kind regards
Bart

David Moransays:

>> the midrange and tweeter are even less omnidirectional so they are sending their sound into a lens that reflects the sound 180 degrees horizontal.
>> The lenses on the beolab 5 where designed to transmit treble and mid-range at a horizontal angle of 180 degrees

Say what ?? Not by my measurements; not, I believe, in Moulton and LaCarrubba’s minds and goals; and not, I also believe I recall, by BeoLab’s own measurements. I am therefore very surprised to read these assertions. While I was not able to measure a BL5 unit outside as I usually do with loudspeakers to analyze horizontal radiation pattern, I did do a close 360deg normalized diff by angle of them in situ (temporally averaged) at Moulton’s studio, and the resultant very nearly equi-omni horizontal radpat curves appeared in SensibleSound magazine 105 (fall 2005) and the Boston Audio Society Speaker newsletter v27n3 the same month. So I believe your description is mistaken.
The BL5 also does not sound at all like a forward-radiating design, even a broad 180deg one.

Serious and uniform refinement of broad (or not broad) wideband beamforming is a extraordinarily important achievement, and I am psyched to audition the BL90. As a longtime owner of Allison and dbx Soundfield designs (as you may or may not know, the former using unique drivers that put out nearly as much sound sideways as forward, the latter the first successful smooth beamforming phased array), I believe I well know how the BL90 will present in its various horizontal radpats. But the precision of your achievement appears to represent a major step forward.

David Moransays:

Yes, I saw the polar maps some time ago, which is why I am so psyched to audition and measure the BL90.

Also yes, I cannot fully believe the BL5 measurement shown. Not only does it not seem to correlate with how the BL5 sounds but also it does not seem plausible that 1k (say) directly behind (-180deg) would be down in level so much. A ~1′ wavelength and that size of lensed horn?

Quite aside from its difference (not huge, but significant enough) from my more casual measurement:

0deg = axis, the flat line; -180deg = the lowest curve, down ~8dB, and usually much less, from say 150Hz to above 10k. There is no reason I can see that the lensed horn would shadow and attenuate the 1k rearward output level quite as BL depicts. If I am reading it right. Mine shows 1k down ~5dB directly behind; yours is quite a bit greater (quieter) than -9dB, is that correct?

Some of this is BL wording and pitch. A design that, broadband, is fully half-loud (-9dB) directly behind (-180deg) is not one I and many other investigators would describe as ‘forward half-omni’.

The production horizontal radpats of Mark Davis’s dbx Soundfield designs (1980s) may interest you guys if you do not know them, phased arrays from back in the day without modern DSP:

I would guess that the differences in the results of your measurement and the one I’m showing is primarily due to the measurement room and windowing. The directivity plot that I’m showing was measured in the Cube using an approximately 23 ms window length – so, a pseudo-anechoic measurement. I wouldn’t be surprised to find a different off-axis level than the one we measure in a non-anechoic environment, but predicting the difference would be impossible due to the influence of the loudspeaker’s directivity and its reflective surroundings, both of which are, of course, frequency dependent. Of course, it is even possible that a measurement at a specific location in a specific room may even result in a lower reading than an anechoic one, due to destructive interference.

Personally, I try to avoid general terms to describe directivity, although sometimes it’s unavoidable when trying to name or explain a concept. So, things like “forward half-omni” or even “wide”, “narrow” or “omni” give some indication, but not really enough information to be useful. So, I prefer to stick with the plots and let the reader sort out whether it’s narrow or wide… For the BeoLab 90, I have seen some people writing that it has a “cardioid” directivity, but this is incorrect. There was no attempt to use the directivity control to emulate a specific pre-defined polar pattern. (As a side note: I had a similar problem recently when I was shopping for a new camera lens and was able on a website to narrow my search results between “wide zooms”, “medium zooms” and “telephoto zooms” – I knew which lens I wanted, I just didn’t know which category it would fit into – which probably means that the category names aren’t descriptive enough.)

Thanks for the link to the Mark Davis paper. I’ve never read it – but I’m sure that some of my colleagues have done so. I’ll put it on the reading list for my summer vacation. I’m particularly curious about the last reference in the bibliography…

cheers
-geoff

David Moransays:

Yeah, I am sure environment accounts for a lot, and as I said I was not outdoors the way I normally measure.

But mine are still delta-only, with the mike spatially averaging within the same vertical slice at the same close distance and the BL5 cabinet turned per measurement, … so environmental loading was held constant (and then subtracted). I would therefore expect more similarity, but would have to see the B&O -180deg ones in detail and also averaged a little bit to comment with savvy.

Anyway, interesting. I am not one who trusts hi-rez measurements of loudspeakers generally in any case, hearing-irrelevant as they demonstrably are most of the time (I was invited to give an NYC AES convention panel presentation on this a few years ago). But I would never say that B&O do not know how to measure.

Point taken about beam descriptions, at least when imprecise. What we want is uniformity, following choice of radpat type in the first place. This was known to many well before Toole (and indeed before Davis as a grad student in 1972), although the Toole/Olive blind work of quantification and preference re hor radpat is simply invaluable.

Anyone who said ‘cardioid’ is a prizewinning idiot and ought not to be writing about audio or anything else. Of course, some say that of planar designs, where it’s not really true either, although it’s less false.

My 30yo Stereo Review article was just a partial look at unacknowledged problems w/ 2-way designs (the now widely recognized trades at the crossover among frequency point, driver width and beaming, direct output, and power output, meaning for the listener trades among imaging tightness, honky MT balance, and listening distance), which back then some were saying were inherently superior or something for bogus reasons maybe having to do w/ “phase”. As an Allison owner and longtime dbx engineering colleague of Davis, I knew better, and thought an article was due. At dbx we also were finishing up the voicing of the flagship Soundfield design, comparing with the new Allison IC20 and the Snell A3, which have about as different hor radpats as one could conceive, so I mentioned that.

I’ve been thinking more about the differences between your directivity measurements and “ours”.

If I understand your method correctly, you put a loudspeaker on a turntable in a “normal” room (whatever that means) and you place the microphone at a location that is some distance from, and on-axis to the loudspeaker. You then rotate the loudspeaker and repeat the measurement at various angles, normalising each measurement to the on-axis measurement.

If you do this, then it seems to me that you’ll experience changes in the constructive and destructive interference pattern between the direct sound, and the various surface reflections at the microphone location. Of course, in a real room, this method probably works as well as is possible – but I think that you’ll measurement something different than would be measured in an anechoic environment.

The problem makes me think back to some of the early papers I’ve seen on pole-zero representations of room acoustics. One of the main messages of those was that, as you move either the loudspeaker or the microphone (or both) in a room, a room’s poles will not move, but the zeros will.

I’m curious as to whether you’ve used your technique to measure the directivity of a loudspeaker, then moved the entire setup to a different location in the same room (or a different one) and repeated the process. If you have done this? And if so, how different are the results?

Cheers
-geoff

David Moransays:

I have done it only once or twice, when weight and/or weather prevented taking the cabinet outside.

In this case, at Moulton’s studio, no industrial turntable, the mike was handheld perhaps a foot away and with the precision pro dbx RTA1 set to continuous averaging and the speaker playing p/n, I moved the mike slowly up and down vertically along the notional axis, from tweeter height to floor and back, a couple of times, moving to and fro (in and out) hardly at all and sideways left-right perhaps a couple of inches. (With the dbx RTA1’s temporal averaging, none of that matters much; the data are altogether repeatable.) I stored the result in memory. I repeated this identically with the cabinet turned -15, -30, -45, -60, -90, -135, and then -180deg or directly behind. One direction only, symmetry assumed. No variables (interferences) change except for cabinet orientation.

Then I subtract the 0deg axis measurement from all of the others, also from itself to produce a straight line (basline). The resultant splayed set of FR curves by angle is what I published in my BL5 review sent to you.

I have done this only one other time and it was during iterative crossover change voicing with a designer, not for publication.

If I were to do this as a regular method instead of my usual outdoors one (measuring at the same angles but moving the mike up/down, side/side within a cubic foot or two in space at seated ear height 7-8′ away), yes, I would compare the two delta sets, lazy indoors vs laborious outdoors, to ensure that they were very close. Obviously you must diff-normalize in order to view hor radpat changes only.

I should have done that QC check for the BL5 review, actually, for complete method QC hygiene.